Echo canceling method and apparatus in a communication device

ABSTRACT

A bidirectional communication device for transmitting and receiving communications signals in a system which presents a transmission path for conducting a communications signal from the device, a reception path for conducting a communications signal to the device, and an echo path which conducts echo signals from the transmission path to the reception path, the device having an echo canceler connected between the transmission path and the reception path for minimizing echo signals in the transmission path, the echo canceler comprising an adaptive filter for filtering the communications signals from the device according to a filter characteristic having a plurality of filter coefficients, wherein a first group of the coefficients, constituting less than all of the coefficients, has finite filter coefficient values and the remaining ones of the plurality of coefficients have values of zero.

This application is a Continuation of application Ser. No. 08/568,843,filed Dec. 7, 1995, which application is incorporated herein byreference.

BACKGROUND OF THE INVENTION

The present invention relates to communication devices such astelephones, and particularly cordless telephones.

The use of adaptive filtering technology to effect echo cancellation inconventional telephone devices is already known in the art. Conventionalecho cancellation techniques are described, for example, byMesserschmitt, in a paper entitled Echo Cancellation in Speech and DataTransmission, IEE JOURNAL ON SELECTED AREAS IN COMMUNICATION, Vol.SAC-2, No. 2, March, 1984, 283-296.

Typically, a telephone set is connected by a single wire pair to afour-wire path forming part of the telephone utility. The four-wire pathconsists of two wires which conduct communication signals in onedirection and two wires which conduct signals in the opposite direction.In order to prevent feedback in such a system, the two wires of thetelephone set are connected to the four-wire path by a device known as ahybrid which is intended to provide signal isolation between the twowire pairs of the four-wire path.

A hybrid does not, however, prevent echos from being propagated in aloop between two connected telephone sets and when, in conventionaltelephone equipment, such an echo has a very long delay, as would beexperienced in the case of connections via satellites, it is alreadyknown to use echo cancelers in the four-wire path near a telephone set.Such cancelers can be constituted, for example, by a finite impulseresponse (FIR) filter which, as is known, consists of a tapped delayline whose taps are connected to multipliers that multiply the signal ateach tap by a filter coefficient, all of the multiplied signals beingsummed to produce a replica of the echo signal which is to be canceled.

It has further been found that echos will also occur in the hybriditself due to inherent impedance mismatches associated with the hybrid.In a conventional telephone set, i.e. a telephone set in which thehandset is connected to the telephone body by a wire, the echo delaythrough the hybrid is of such short duration that it is not noticeableto the user.

However, in the case of digital cordless telephones (DCTs) communicationsignals are transmitted between the handset and the base station inpackets, each of which may contain about four milliseconds of audiodata. Because of the resulting built-in delay in the transmission ofaudio data, an echo signal that passes through the hybrid will beperceivable to the user. This problem exists in the general class ofTime Division Multiple Access (TDMA) systems.

SUMMARY OF THE INVENTION

It is an object of the present invention to cancel echos which areproduced at a telephone set.

A further object of the invention is to perform adaptive echocancellation in a manner which minimizes required computation resources.

A further object of the invention is to perform adaptation of an echocanceling filter to existing conditions at appropriate times.

These and other objects are achieved by a bidirectional communicationdevice for transmitting and receiving communications signals in a systemwhich presents a transmission path for conducting a communicationssignal from the device to the telephone network, a reception path forconducting a communications signal to the device from the telephonenetwork, and an echo path which conducts echo signals from thetransmission path to the reception path, the device having an echocanceler connected between the transmission path and the reception pathfor minimizing echo signals in the reception path, the echo cancelercomprising an adaptive filter for filtering the communications signalsfrom the device according to a filter characteristic having a pluralityof filter coefficients, wherein a first group of the coefficients,constituting less than all of the coefficients, has finite filtercoefficient values and the remaining ones of the plurality ofcoefficients have values that are approximately zero.

BRIEF DESCRIPTION OF THE DRAWING

FIG. 1 is a block diagram showing a telephone base station equipped withan echo canceler according to a preferred embodiment of the invention.

FIG. 2 is a block diagram illustrating the principle of an adaptivefilter employed in the embodiment of FIG. 1.

FIG. 3 is a flow diagram of an echo canceler adaptation procedureaccording to the invention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 shows the basic components of a DCT base station provided with anecho canceler in the form of an adaptive filter. The fundamentalcomponents of the base station include a TDMA modem 2 and a CODEC whichis constituted by a digital-analog converter section 4 and ananalog-digital converter section 6. Converter sections 4 and 6 arecoupled to a telephone line via a hybrid 10 which is here assumed to bean echo channel via which echo signals are propagated from convertersection 4 to converter section 6. Modem 2 is a TDMA device. This deviceshares communication resources by first configuring itself as atransmitter and sending data in burst to the handset. Then modem 2reconfigures itself as a receiver and receives data in bursts from thehandset. Because modem 2 must gather speech data into bursts, and sometime is required to reconfigure between the transmitter configurationand the receiver configuration, TDMA systems have an unavoidable delay.Hybrid 10 is constructed to couple converter section 4 to an outgoingwire pair of the telephone line, and to couple converter section 6 to anincoming wire pair of the telephone line. When an impedance mismatchexists at hybrid 10, a portion of the communication signals fromconverter section 4 will be conducted through hybrid 10 to the input ofconverter section 6, where they will be converted into digital form andthen broadcast by modem 2 to the handset. Because of the inherent delayassociated with the TDMA transmission of voice signals to and from thehandset, an echo will be perceived by the user and will be found to beobjectionable.

Such echo can be effectively neutralized by the connection of anadaptive filter 12 between the path from modem 2 to converter section 4and the path from converter section 6 to modem 2. Specifically, theinput of adaptive filter 12 is connected to the input of convertersection 4 and the output of filter 12 is connected to a summing element14 which also receives, as a second input, a signal appearing at theoutput of analog-digital converter section 6. The output of summingelement 14 provides a signal from which the echo propagated throughhybrid 10 is canceled, or at least minimized.

Thus, in the embodiment shown in FIG. 1, the echo canceler isconstituted essentially by filter 12 and summing element 14.

Adaptive filter 12 may be a finite impulse response (FIR) filter whoseoperation is controlled by an LMS algorithm, also known in the art as astochastic gradient algorithm. However, any other suitable algorithm canbe employed and an adaptive filter other than a FIR filter may beutilized. One advantage of operating such a filter on the basis of a LMSalgorithm is that such an algorithm requires less computation resourcesthan other types of algorithms. In existing and proposed DCTs,computation resources are limited.

A typical example of adaptive filter 12 can be represented in the mannershown in FIG. 2. The adaptive filter is composed essentially of a seriesarrangement of delay lines 20, with the first delay line in the seriesarrangement being coupled to a signal input 22. Input 22 and the outputof each delay line 20 are each connected to one input of a respectivemultiplier 24. The other input of each multiplier 24 is connected toreceive a signal representing a respective filter coefficient a₀, a₁, a₂. . . a_(n−1). Thus, this delay line is provided with n taps and can becontrolled by n filter coefficients.

The outputs of all of the multipliers 24 are connected to a summingelement 26 which supplies a filter output signal at output 28.

Filters of the type described above generally have a significant numberof taps and the conventional practice in the art is to calculate afinite filter coefficient for each tap. For example, existing FIRadaptive filters may have of the order of 20 or more taps. Adaptivecontrol of all of the taps of such a filter requires significantcomputational resources. However, Applicants have determined that ahighly effective echo cancellation can be achieved by controlling thefilter coefficients for only a limited number of the taps and settingall of the other filter coefficients to a value of zero. By way ofexample, in the case of an adaptive filter which contains 20 taps, ithas been found that the assigning of appropriate non-zero filtercoefficients to six of the taps will produce a significant echoreduction, while if filter coefficient values are calculated for onlyfour to six of the taps, significant echo reduction cannot be achieved.

When filter coefficients are computed for only a limited number of taps,the required computation resources are reduced.

In further accordance with the invention, the filter coefficients forappropriate taps are selected periodically by the operation of a filtercontrol 16, as shown in FIG. 1. Filter control 16 may be implemented bya digital controller provided in the base station under control ofprogramming stored in a memory also forming part of the base station.

In order to further accommodate the limited computational resourceswithin the base station, filter adaptation is not performed during thecourse of a telephone conversation, it being assumed that the conditionswhich would affect operation of adaptive filter 12 will not vary on ashort term basis. Thus, according to the invention, adjustment of theadaptive filter coefficients is performed during dialing, i.e. whenevera call is placed from the telephone set. Studies have shown that themost desirable time for adjusting the filter coefficients is in thesilent interdigital intervals between actuations of one of the keypadsassociated with the telephone set, there typically being one keypad onthe base station and one keypad on the handset.

Adjustment of the filter coefficients is not effected at the time of anincoming call, it being assumed that the conditions influencing echobehavior have not changed significantly since the last outgoing call.

FIG. 3 is a block diagram showing one embodiment of a procedureaccording to the invention for setting selected echo coefficients in anadaptive filter of the echo canceler.

Step 101 is initiated by the release of a keypad button and includesestablishing a certain delay period and then turning off the microphoneat the telephone instrument handset.

In step 102 the microphone is monitored and if no sounds are heard forfive milliseconds, a decision block in step 103 triggers a step 104 inwhich the transmission path is blocked and transmission of anysubsequent key inputs is delayed.

Then, in steps 105-110, there is performed a cyclic process whichincludes training operations (steps 105, 108) and wait periods andsilence detection (steps 106, 107, 109 and 110). The training operationsneed not be performed after a first keypad actuation. In fact, it ispreferable that the training operations be carried out after inputtingof the second digit which is inputted. During calls when only one digitis inputted, for example when the operator is called, the trainingprocess may not be reliable.

The training signal is applied by using switch 19 to disconnect themodem 2 and connect to noise generator 18 in steps 105, 108 and may beconstituted by filtered white noise generated by noise generator 18,using uniform distributed random sequences applied through a low passfilter as an input. The training signals can be given a desiredbandwidth, it having been found that larger bandwidth signals produceimproved echo reduction.

Based on experiments which have been conducted to date, it has beenfound that a 4th order Butterworth low pass filter with a three dBbandwidth of 3.6 kHz provides a good signal for training purposes.

The training signal level can vary over a certain range and should belarge enough to provide a satisfactory signal-noise ratio and smallenough to avoid erroneously triggering various telephone functions. Ithas been found preferable that the training signal level be in the rangeof −25 to −15 dBm, corresponding to a signal voltage in the order of45-150 vMrms.

In addition, for a LMS algorithm, the step size of filter coefficientchanges affects convergence performance. The optimum step size varieswith training signal levels and the following relationship between thevarious input signal levels and optimum step size has been found toexist.

Input signal (vpp) 4.0 2.0 1.0 0.5 0.25 Step size (v) 0.0078 0.03130.125 0.5 2.0

The silence detection performed in steps 102, 106 and 109, serves thepurpose of detecting whether the telephone line is silent, except forthe training signal, before, during and after a training phase. Thiswill assure that training is performed under conditions when no incomingsignal is on the telephone line. During this step, the line isconsidered to not be silent if any sample voltage larger than 5 mV isdetected during the 5 ms interval.

It has been found that the training period should be at least 30 ms andthat a training period of 100 ms allows filter coefficient values to beaccurately determined.

As indicated by the flow diagram of FIG. 3, it is preferable that thetraining period be divided into two subperiods each having a duration ofessentially 35 ms, with an interval of 45 ms between them to assure thatthe training signal will not have any adverse effect on telephonenetworks.

The training process involves the use of a known algorithm, such as aLMS algorithm to determine the appropriate filter coefficients at alltaps. This represents the operation performed in step 111 of FIG. 3. Theprocess performed in steps 111 through 115 includes sending filterednoise signals two times for a period of 5 ms each (steps 111 and 113)and examining the output of the adaptive filter if all taps wereadjusted (step 112), and the output of the adaptive filter if six filtercoefficients have been set to finite values and the remainder to a zerovalue (step 114). If the average residual error in the case when allfilter coefficients have been set is not more than 10 dB below the levelof the input to the canceler, or in the case of truncated tap control,i.e. control of a limited number of filter coefficients, the averageresidual error is not more than 6 dB below the level of the cancelerinput signal, the training process is aborted, the current filtercoefficients are canceled, the echo canceler is disabled, and theprevious filter coefficients are reloaded (step 115).

Filter adaptation techniques and algorithms are already known in the artand the present invention proposes to utilize any suitable one of thesethe techniques, modified to identify a selected number of the mostsignificant filter coefficients which have the largest values. Thelocations of the taps associated with these coefficients can, of course,vary from one training operation to another.

The solution proposed by the present invention is based on theassumption that only small variations will occur in loop characteristicsduring a single telephone call. Although there is evidence that thisassumption is generally correct, it has not yet been verified bysufficiently exhaustive testing.

While the description above refers to particular embodiments of thepresent invention, it will be understood that many modifications may bemade without departing from the spirit thereof. The accompanying claimsare intended to cover such modifications as would fall within the truescope and spirit of the present invention.

The presently disclosed embodiments are therefore to be considered inall respects as illustrative and not restrictive, the scope of theinvention being indicated by the appended claims, rather than theforegoing description, and all changes which come within the meaning andrange of equivalency of the claims are therefore intended to be embracedtherein.

What is claimed:
 1. A method for setting the initial values ofcoefficients of an adaptive filter used for minimizing echo signals in areception path of a digital cordless telephone said method beingperformed between the second and third digits dialed after the telephonegoes off the hook, the method comprising: supplying a test signalthrough the echo path; monitoring the response of the device to the testsignal; separating filter coefficients into a first and second group;selecting the coefficients of the first group based on initial values ofgreatest magnitude; and zeroing coefficients of the second group havinginitial values of lesser magnitude.
 2. The method of claim 1 comprisingdetermining the initial values for all of the adaptive filtercoefficients at the start of a conversation.
 3. The method of claim 1comprising selecting and no longer adapting the first group ofcoefficients having initial values of greatest magnitude, constitutingless than all of the coefficients and for retaining the initial valuesof the first group of coefficients throughout the conversation.
 4. Themethod of claim 1 wherein determining initial values for all of theadaptive filter coefficients comprises monitoring echo propagation alongthe echo path and adjusting the filter coefficient values in response tothe monitored echo signal propagation.
 5. The method of claim 1 whereinsaid digital cordless telephone comprises a handset from which signalsare transmitted in the form of packets which are transmitted at spacedtime intervals.
 6. The method of claim 1 wherein said adaptive filter isa digital finite impulse response filter.
 7. The method of claim 1wherein said adaptive filter produces an output signal constituting apredicted echo signal.
 8. The method of claim 1 wherein said digitalcordless telephone comprises both a handset and a base station.
 9. Themethod of claim 1 wherein the digital cordless telephone comprises akeypad provided with a plurality of numeric keys and said method isperformed in response to actuation of a selected number of keys.
 10. Amethod for setting the initial values of coefficients of an adaptivefilter used for minimizing echo signals in a reception path of a digitalcordless telephone said method being performed between dialed digitsafter the telephone goes off the hook, the method comprising: supplyinga test signal through the echo path; and monitoring the response of thedevice to the test signal.
 11. The method of claim 10 comprisingdetermining the initial values for all of the adaptive filtercoefficients at the start of a conversation.
 12. The method of claim 10wherein said setting the initial values of the coefficients comprisesselecting and no longer adapting the coefficients and for retaining theinitial values of the coefficients throughout the conversation.
 13. Themethod of claim 10 wherein determining initial values for all of theadaptive filter coefficients comprises monitoring echo propagation alongthe echo path and adjusting the filter coefficient values in response tothe monitored echo signal propagation.
 14. The method of claim 10wherein said digital cordless telephone comprises a handset from whichsignals are transmitted in the form of packets which are transmitted atspaced time intervals.
 15. The method of claim 10 wherein said adaptivefilter is a digital finite impulse response filter.
 16. The method ofclaim 10 wherein said adaptive filter produces an output signalconstituting a predicted echo signal.
 17. The method of claim 10 whereinsaid digital cordless telephone comprises both a handset and a basestation.
 18. The method of claim 10 wherein the digital cordlesstelephone comprises a keypad provided with a plurality of numeric keysand said method is performed in response to actuation of a selectednumber of keys.
 19. The method of claim 10 wherein the method isperformed between second and third digits dialed after the telephonegoes off hook.